Fixed IP networks were originally designed to carry “best effort” traffic where the network makes a “best attempt” to deliver a user packet, but does not guarantee that a user packet will arrive at the destination. IP networks need to support various types of applications. Some of these applications have Quality of Service (QoS) requirements other than “best effort” service. Examples of such applications include various real time applications (IP telephony/voice, video conferencing), streaming services (audio or video), or high quality data services (browsing with bounded download delays). Recognizing these QoS requirements, the Internet Engineering Task Force (IETF), which is the main standards body for IP networking, standardized a set of protocols and mechanisms that enable IP network operators to build QoS-enabled IP networks. But these protocols and mechanisms where designed with fixed, wire-line networks in mind. New and different challenges face IP communications in mobile, wireless communication networks.
Quality of service is important for providing end users with satisfying service. The efficient use of the radio resources is also important to ensure maximum capacity and coverage for the system. Quality of service can be characterized by several performance criteria such as throughput, connection setup time, percentage of successful transmissions, speed of fault detection and correction, etc. In an IP network quality of service can be measured in terms of bandwidth, packet loss, delay, and jitter.
Consider for example an IP telephony session between User-A and User-B where User-A accesses an IP backbone through a local access, mobile communications network. In wireline communications, a local access network is often a Public Switched Telephone Network (PSTN or an Integrates Services Digital Network (ISDN. But for communications involving a mobile radio, the local access network must include a radio access network. Example mobile communication networks include the Global System for Mobile communications (GSM) or the Universal Mobile Telecommunications System (UMTS) network. User-B is similarly connected to the IP network through a local access network, and both users may not use the same type of access network. The IP backbone network includes a number of IP routers and interconnecting links that together provide connectivity between the IP network's ingress and egress points and thereby make two party communication possible.
As far as the users are concerned, the perceived quality of service depends on the service provided both in the local access networks and on the IP backbone network Of particular interest is the specific case where at least one of the access networks is a mobile communications network like a UMTS or GSM/GPRS networks. The radio interface in such a network is the most challenging interface in the communication in terms of delivering a particular quality of service.
An objective of third generation mobile communications systems, like Universal Mobile Telephone communications System (UMTS), is to provide mobile radios with the ability to conduct multimedia sessions where a communication session between users may include different types of media. Perhaps the most important medium to support in multimedia sessions is voice. There is a need for more resource-efficient, packet-based conversational (e.g., voice) multimedia services. Although the idea of conversational IP services is desirable, a practical implementation of a conversational IP service requires overcoming several technical hurdles before the idea becomes a commercial reality. Conversational IP services should deliver high speech quality both in terms of fidelity and low delay. Connection set up and service interaction times should be reasonably fast. Indeed, packet-based voice service should be comparable to circuit-switched traditional voice telephony. The radio spectrum must be used efficiently. Services must cover a wide geographic area and be able to service roaming users. Because voice is only one component in a multimedia session, it should be established and disconnected independently from the session.
There are three distinct information flows to be considered for a conversational IP voice service. Each flow affects the overall performance of the IP voice service. For example, the voice media flow is crucial when it comes to providing high speech quality. The session control flow is important when it comes to service set-up times/delays, and the media control protocol is primarily used to monitor media flows and provide information allowing the synchronization of different media flows.
In addition, network operators must be able to provide the conversational IP service at a reasonable cost. Although fixed, wire-line access networks like LAN's permit over-provisioning, wireless networks cannot afford that luxury because of limited radio bandwidth and the need to support user mobility. An objective of the present invention is to provide an efficient way to transport these three information flows.
The three information flows have different needs and characteristics. Quality of Service (QoS) parameters of special importance to session signaling include bit rate, delay, and priority. Session signaling is mostly low volume with a small average bandwidth demand. But the transmission rate still needs to be fairly high to reduce delay. Delay time is also influenced by bearer-handling delays and retransmissions over the air interface. Under heavy load conditions, session signaling should get priority. The session signaling should not be put on a bearer that carries a large volume of user data because the session signaling can not then be prioritized above the user data, resulting in undesired delays. Therefore, the session signaling may be transported using a separate packet data “context” between the mobile terminal and a packet-based access network with an interactive class of QoS. This session signaling packet data context is supported by a dedicated bearer with an interactive class of QoS. Logical connections, like a packet data context, a radio access bearer, a radio bearer, etc., are generally referred to as “bearers.”
The voice media packets carry regularly-generated voice samples, e.g., every 20 ms, and each packet has a relatively small payload size. Those voice packets must be received by the remote terminal with the same timing, e.g., every 20 ms, in order to have a reasonable voice quality. Both objectives are met in accordance with another aspect of the present invention where the voice media flow is transported using a separate packet context between the mobile terminal and a packet-based access network with a conversational class of QoS. The voice media packets are supported by a dedicated bearer with a conversational class of QoS characteristics.
Compared to voice media packets, voice media control message packets are considerably larger in size and are sent less frequently, e.g., every few seconds. But there is no strict delay or jitter requirement like there is for voice packets. For media control message packets, a conversational radio access bearer would waste radio resources. Alternatively, transporting the media control message packets on the same radio access bearer as the voice packets would delay the voice packets causing a disruption in speech. To mitigate this problem requires allocating more resources to a single bearer than would be needed for two bearers. Thus, the media control signaling may be transported using a separate packet data “context” between the mobile terminal and a packet-based access network with an interactive class of QoS. This media control signaling packet data context is supported by a dedicated bearer with an interactive class of QoS characteristics.
Thus, in a preferred example embodiment, each of these three information flows required for a conversational IP voice service is allocated its own bearer and packet data context with a QoS class tailored to the characteristics of each flow. In this way, high quality, packet-based voice service can be provided with radio resources being efficiently allocated in accordance with the particular needs for the different information flows.
In a second example embodiment, the session signaling and the voice media control messages share a single radio bearer with an interactive class of QoS characteristics. As in the first embodiment, the voice media packets are supported by a dedicated bearer with a conversational class of QoS characteristics. Because the session signaling load normally is heavy during the session setup and the voice media control message load normally picks up after the session is setup, one interactive QoS bearer supports both the session setup signaling and the voice media control signaling. Although there may be some delays whenever there is overlapping session signaling and voice media control signaling require transmission at the same time, those delays may be an acceptable tradeoff to reduce the number of bearers by one.